ex_webrtc
An Elixir implementation of the W3C WebRTC API (by elixir-webrtc)
sip
SIP to WebRTC bridge for LiveKit (by livekit)
The number of mentions indicates the total number of mentions that we've tracked plus the number of user suggested alternatives.
Stars - the number of stars that a project has on GitHub. Growth - month over month growth in stars.
Activity is a relative number indicating how actively a project is being developed. Recent commits have higher weight than older ones.
For example, an activity of 9.0 indicates that a project is amongst the top 10% of the most actively developed projects that we are tracking.
Stars - the number of stars that a project has on GitHub. Growth - month over month growth in stars.
Activity is a relative number indicating how actively a project is being developed. Recent commits have higher weight than older ones.
For example, an activity of 9.0 indicates that a project is amongst the top 10% of the most actively developed projects that we are tracking.
ex_webrtc
Posts with mentions or reviews of ex_webrtc.
We have used some of these posts to build our list of alternatives
and similar projects. The last one was on 2024-01-04.
sip
Posts with mentions or reviews of sip.
We have used some of these posts to build our list of alternatives
and similar projects. The last one was on 2024-01-04.
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Show HN: Bring phone calls into the browser (sip-to-WebRTC)
Hi Sean! I'm guessing this is derives as part of the recent collaboration with LiveKit to add SIP support, right?
I had been looking at https://github.com/livekit/sip the other day, so it hasn't been much of a surprise to now see this here. Cool integration!
What are some alternatives?
When comparing ex_webrtc and sip you can also consider the following projects:
obs-studio - OBS Studio - Free and open source software for live streaming and screen recording
str0m - A synchronous sans I/O WebRTC implementation in Rust.
webrtc-for-the-curious - WebRTC for the Curious: Go beyond the APIs